Interpolation Filters.  Write a Matlab program to read and upsample the linked v

Interpolation Filters.  Write a Matlab program to read and upsample the linked voice_samp_8k.wav file (below — sampled at Fs=8000 samples per second) by an integer factor L (you can choose, probably in the range from 3 to 7 or so).  Then use interpolation filtering to produce the following outputs and listen to them and plot short segments (~50-100 samples) of these outputs:  (i) upsample only with no interpolation filtering, (ii) hold interpolation implemented directly as a causal FIR filter (i.e., using filter() function), (iii) linear interpolation implemented directly as a causal FIR filter (i.e., using filter() function), and two other designed FIR lowpass interpolation filtering with (iv) shorter (~20-200) FIR filters and (v) longer (~200-2000) FIR filters that you design with fir1().  Verify that your program is doing what you want by looking at plots of the various outputs.
Matlab Functions that will help you:  soundsc() to hear the outputs, audioread() to read in the 12.5 sec speech file, kron() for single line upsampling, fir1() from our in-class examples, filter() to produce filtered versions of your upsampled signal, freqz() to assess frequency response for the plots, and plot() to plot.